Method and apparatus for processing an audio signal, audio decoder, and audio encoder to filter a discontinuity by a filter which depends on two fir filters and pitch lag

ABSTRACT

A method is described that processes an audio signal. A discontinuity between a filtered past frame and a filtered current frame of the audio signal is removed using linear predictive filtering.

CROSS-REFERENCE TO RELATED APPLICATIONS

This application is a continuation of copending U.S. patent applicationSer. No. 15/412,920, filed Jan. 23, 2017, which is incorporated hereinby reference in its entirety, which in turn is a continuation ofcopending International Application No. PCT/EP2015/065219, filed Jul. 3,2015, which is incorporated herein by reference in its entirety, andadditionally claims priority from European Application No. 14178821.6,filed Jul. 28, 2014, which is also incorporated herein by reference inits entirety.

BACKGROUND OF THE INVENTION

The present invention relates to the field of audio signals, morespecifically to an approach for processing an audio signal including aplurality of audio frames, wherein discontinuities between consecutivefiltered audio frames are reduced or omitted.

In the field of audio signal processing, an audio signal may be filteredfor various reasons, e.g., a long-term prediction filter may be used inan audio signal encoder, to attenuate or even suppress completely a setof harmonics in the audio signal.

The audio signal includes a plurality of audio frames, and the framesare filtered using the long-term prediction filter. When considering twoconsecutive frames of an audio signal, a past frame and a current frame,a linear filter H(z) having a set of parameters c is used for filteringthe audio signal. More specifically, the past frame is filtered with thefilter H(z) using a first set of parameters c₀ which will produce aso-called filtered past frame. The current frame is filtered with thefilter H(z) using a set of parameters c₁ which will produce a filteredcurrent frame. FIG. 1 shows a block diagram for processing consecutiveframes of an audio signal in accordance with a known approach. An audiosignal 100 including a plurality of audio frames is provided. The audiosignal 100 is supplied to a filter block 102 and a current frame n ofthe audio signal 100 is filtered. The filter block, besides the audiosignal 100, receives a set of filter parameters c_(n) for the currentframe of the audio signal. The filter block 102 filters the currentframe n of the audio signal and outputs a filtered audio signal 104including consecutive filtered frames. In FIG. 1 , the filtered currentframe n, the filtered past frame n−1 and the filtered second last framen−2 are schematically depicted. The filtered frames are schematicallyrepresented in FIG. 1 with respective gaps therebetween forschematically indicating a discontinuity 106 a, 106 b that may beintroduced by the filtering process between the filtered frames. Thefilter block 102 causes filtering of the frames of the audio signalusing respective filter parameters c₀ and c₁ for a past frame n−1 and acurrent frame n. In general, the filter block 102 may be a linear filterH(z), and one example for such a linear filter H(z) is the abovementioned long-term prediction filterH(z)=1−g·z ^(−T)

where the filter parameters are the gain “g” and the pitch lag “T”. In amore general form, the long-term prediction filter can be described asfollows:H(z)=1−g·A(z)·z ^(−T)

where A(z) is a FIR filter. A long-term prediction filter may be used toattenuate or even suppress completely a set of harmonics in an audiosignal. However, there is a high probability of introducing adiscontinuity 106 a, 106 b (see FIG. 1 ) between the filtered past framen−1 and the filtered current frame n when using such a long-termprediction filter and when the past frame filter parameters c₀ aredifferent from the current frame filter parameters c₁. Thisdiscontinuity may produce an artifact in the filtered audio signal 104,for example a “click”.

Consequently, in view of the above described problems with the filteringof consecutive frames resulting in discontinuities which, in turn, mayproduce undesired artifacts, a technique is needed that removes apossible discontinuity. Several known approaches dealing with theremoval of a discontinuity of filtered frames of an audio signal areknown in the art.

In case the linear filter H(z) is a FIR filter, the current frame isfiltered with the filter parameters c₁ of the current frame forproducing a filtered current frame. In addition, a beginning portion ofthe current frame is filtered with the filter parameters of the pastframe c₀ for producing a filtered frame portion, and then an overlap-addor cross-fade operation is performed over the beginning portion of thefiltered current frame and the filtered frame portion. FIG. 2 shows ablock diagram of such a conventional approach for processing consecutiveaudio frames for removing a discontinuity. When compared to FIG. 1, thefilter block 102 includes a further processing block 108 for performingthe overlap-add or cross-fade operation. In the filtered audio signal104, there will be no or a reduced discontinuity between the consecutivefiltered frames, as is schematically indicated in FIG. 2 showing theconsecutive filtered frames n, n−1 and n−2 without the gaps of FIG. 1 .

In other known approaches, the filter H(z) may be a filter having arecursive part, for example an IIR filter. In such a case, the approachas described above with regard to FIG. 2 is applied on asample-by-sample basis. In a first step, the processing starts with thefirst sample of the beginning portion of the current frame n beingfiltered with the filter parameters c₀ of the past frame n−1 yielding afirst filtered sample. The sample is also filtered with the filterparameters c₁ of the current frame n producing a second filtered sample.Then, the overlap-add or cross-fade operation is performed based on thefirst and second filtered samples which yields the corresponding sampleof the filtered current frame n. Then the next sample is processed andthe above steps are repeated until the last sample of the beginningportion of the current frame n has been processed. The remaining samplesof the current frame n are filtered with the filter parameters c₁ of thecurrent frame n.

Examples for the above mentioned known approaches for removing adiscontinuity from consecutive filtered frames are described, forexample, in U.S. Pat. No. 5,012,517 A in the context of a transformcoder, in EP 0732687 A2 in the context of a speech bandwidth expander,in U.S. Pat. No. 5,999,899 A in the context of a transform audio coder,or in U.S. Pat. No. 7,353,168 B2 in the context of a decoded speechpostfilter.

While the above approaches are efficient for removing the undesiredsignal discontinuities, since these approaches operate on a specificportion of the current frame, the beginning portion, for beingeffective, the length of the frame portion has to be sufficiently long,for example in the case of a frame length of 20 ms, the frame portion orbeginning portion length could be as long as 5 ms. In certain cases,this can be too long, especially in situations where the past framefilter parameters c₀ will not apply well to the current frame and thismay result in additional artifacts. One example is a harmonic audiosignal with fast changing pitch, and a long-term prediction filter thatis designed to reduce the amplitude of the harmonics. In that case, thepitch-lag is different from one frame to the next. The long-termprediction filter with the pitch estimated in the current frame wouldeffectively reduce the amplitude of the harmonics in the current frame,but it would not reduce the amplitude of the harmonics if used inanother frame (e.g. beginning portion of the next frame) where the pitchof the audio signal would be different. It could even make things worse,by reducing the amplitude of non-harmonic-related components in thesignal, introducing a distortion in the signal

It is an object underlying the present invention to provide an improvedapproach for removing discontinuities among filtered audio frameswithout producing any potential distortion in the filtered audio signal.

SUMMARY

According to an embodiment, a method for processing an audio signal mayhave the step of: removing a discontinuity between a filtered past frameand a filtered current frame of the audio signal using linear predictivefiltering.

Another embodiment may have a non-transitory digital storage mediumhaving stored thereon a computer program product for performing a methodfor processing an audio signal, the method having the step of: removinga discontinuity between a filtered past frame and a filtered currentframe of the audio signal using linear predictive filtering, when saidcomputer program is run by a computer.

Still another embodiment may have an apparatus for processing an audiosignal, wherein the apparatus comprises a processor for removing adiscontinuity between a filtered past frame and a filtered current frameof the audio signal using linear predictive filtering, or wherein theapparatus is configured to operate according to a method for processingan audio signal, the method having removing a discontinuity between afiltered past frame and a filtered current frame of the audio signalusing linear predictive filtering.

Another embodiment may have an audio decoder having the above inventiveapparatus for processing an audio signal.

Another embodiment may have an audio encoder having the above inventiveapparatus for processing an audio signal.

The present invention provides a method for processing an audio signal,the method comprising removing a discontinuity between a filtered pastframe and a filtered current frame of the audio signal using linearpredictive filtering.

The linear predictive filter can be defined as

${A(z)} = \frac{1}{\Sigma_{m = 0}^{M}a_{m}z^{- m}}$

with M the filter order and a_(m) the filter coefficients (with a₀=1).This kind of filter is also known as Linear Predictive Coding (LPC).

In accordance with embodiments, the method comprises filtering thecurrent frame of the audio signal and removing the discontinuity bymodifying a beginning portion of the filtered current frame by a signalobtained by linear predictive filtering a predefined signal with initialstates of the linear predictive filter defined on the basis of a lastpart of the past frame.

In accordance with embodiments, the initial states of the linearpredictive filter are defined on the basis of a last part of theunfiltered past frame filtered using the set of filter parameters forfiltering the current frame.

In accordance with embodiments, the method comprises estimating thelinear predictive filter on the filtered or non-filtered audio signal.

In accordance with embodiments, estimating the linear predictive filtercomprises estimating the filter based on the past or current frame ofthe audio signal or based on the past filtered frame of the audio signalusing the Levinson-Durbin algorithm.

In accordance with embodiments, the linear predictive filter comprises alinear predictive filter of an audio codec.

In accordance with embodiments, removing the discontinuity comprisesprocessing the beginning portion of the filtered current frame, whereinthe beginning portion of the current frame has a predefined number ofsamples being less or equal than the total number of samples in thecurrent frame, and wherein processing the beginning portion of thecurrent frame comprises subtracting a beginning portion of azero-input-response (ZIR) from the beginning portion of the filteredcurrent frame.

In accordance with embodiments, the method comprises filtering thecurrent frame of the audio signal using a non-recursive filter, like aFIR filter, for producing the filtered current frame.

In accordance with embodiments, the method comprises processing theunfiltered current frame of the audio signal on a sample-by-sample basisusing a recursive filter, like an IIR filter, and wherein processing asample of the beginning portion of the current frame comprises:

-   -   filtering the sample with the recursive filter using the filter        parameters of the current frame for producing a filtered sample,        and    -   subtracting a corresponding ZIR sample from the filtered sample        for producing the corresponding sample of the filtered current        frame.

In accordance with embodiments, filtering and subtracting are repeateduntil the last sample in the beginning portion of the current frame isprocessed, and wherein the method further comprises filtering theremaining samples in the current frame with the recursive filter usingthe filter parameters of the current frame.

In accordance with embodiments, the method comprises generating the ZIR,wherein generating the ZIR comprises:

-   -   filtering the M last samples of the unfiltered past frame with        the filter and the filter parameters used for filtering the        current frame for producing a first portion of filtered signal,        wherein M is the linear predictive filter order,    -   subtracting from the first portion of filtered signal the M last        samples of the filtered past frame, filtered using the filter        parameters of the past frame, for generating a second portion of        filtered signal, and    -   generating a ZIR of a linear predictive filter by filtering a        frame of zero samples with the linear predictive filter and        initial states equal to the second portion of filtered signal.

In accordance with embodiments, the method comprises windowing the ZIRsuch that its amplitude decreases faster to zero.

The present invention is based on the inventor's findings that theproblems that have been recognized in conventional approaches forremoving signal discontinuities which result in the additional unwanteddistortion mentioned above, are mainly due to the processing of thecurrent frame or at least a portion thereof on the basis of the filterparameters for the past frame. In accordance with the inventive approachthis is avoided, i.e. the inventive approach does not filter a portionof the current frame with the filter parameters of the past frame andthus avoids the problems mentioned above. In accordance withembodiments, for removing the discontinuity, an LPC filter (linearpredictive filter) is used for removing the discontinuity. The LPCfilter may be estimated on the audio signal and therefore it is a goodmodel of the spectral shape of the audio signal so that, when using theLPC filter, the spectral shape of the audio signal will mask thediscontinuity. In an embodiment, the LPC filter may be estimated on thebasis of the non-filtered audio signal or on the basis of an audiosignal that has been filtered by a linear filter H(z) mentioned above.In accordance with embodiments, the LPC filter may be estimated by usingthe audio signal, for example the current frame and/or the past frame,and the Levinson-Durbin algorithm. It may also be computed only on thebasis of the past filtered frame signal using the Levinson-Durbinalgorithm.

In yet other embodiments, an audio codec for processing the audio signalmay use a linear filter H(z) and may also use an LPC filter, eitherquantized or not, for example to shape the quantization noise in atransform-based audio codec. In such an embodiment, this existing LPCfilter can be directly used for smoothing the discontinuity without theadditional complexity needed to estimate a new LPC filter.

BRIEF DESCRIPTION OF THE DRAWINGS

In the following, embodiments of the present invention will be describedwith reference to the accompanying drawings, in which:

FIG. 1 shows a block diagram for processing consecutive frames of anaudio signal in accordance with a conventional approach;

FIG. 2 shows a block diagram of another conventional approach forprocessing consecutive audio frames for removing a discontinuity;

FIG. 3 shows a simplified block diagram of a system for transmittingaudio signals implementing the inventive approach for removing adiscontinuity between consecutive frames of an audio signal at theencoder side and/or at the decoder side;

FIG. 4 shows a flow diagram depicting the inventive approach forremoving a discontinuity between consecutive frames of an audio signalin accordance with an embodiment;

FIG. 5 shows a schematic block diagram for processing a current audioframe in accordance with embodiments of the present invention avoidingundesired distortion in the output signal despite the removal of thediscontinuities;

FIG. 6 shows a flow diagram representing the functionality of the blockin FIG. 5 for generating the ZIR;

FIG. 7 shows a flow diagram representing the functionality of the blockin FIG. 5 for processing the filtered current frame beginning portion incase the filter block comprises a recursive filter, like an IIR filter;and

FIG. 8 shows a flow diagram representing the functionality of the blockin FIG. 5 for processing the filtered current frame beginning portion incase the filter block comprises a non-recursive filter, like a FIRfilter.

DETAILED DESCRIPTION OF THE INVENTION

In the following, embodiments of the inventive approach will bedescribed in further detail and it is noted that in the accompanyingdrawing elements having the same or similar functionality are denoted bythe same reference signs.

FIG. 3 shows a simplified block diagram of a system for transmittingaudio signals implementing the inventive approach at the encoder sideand/or at the decoder side. The system of FIG. 3 comprises an encoder200 receiving at an input 202 an audio signal 204. The encoder includesan encoding processor 206 receiving the audio signal 204 and generatingan encoded audio signal that is provided at an output 208 of theencoder. The encoding processor may be programmed or built to implementthe inventive approach for processing consecutive audio frames of theaudio signal received to avoid discontinuities. In other embodiments theencoder does not need to be part of a transmission system, however, itcan be a standalone device generating encoded audio signals or it may bepart of an audio signal transmitter. In accordance with an embodiment,the encoder 200 may comprise an antenna 210 to allow for a wirelesstransmission of the audio signal, as is indicated at 212. In otherembodiments, the encoder 200 may output the encoded audio signalprovided at the output 208 using a wired connection line, as it is forexample indicated at reference sign 214.

The system of FIG. 3 further comprises a decoder 250 having an input 252receiving an encoded audio signal to be processed by the encoder 250,e.g. via the wired line 214 or via an antenna 254. The encoder 250comprises a decoding processor 256 operating on the encoded signal andproviding a decoded audio signal 258 at an output 260. The decodingprocessor 256 may be implemented to operate in accordance with theinventive approach on consecutive frames that are filtered in such a waythat discontinuities are avoided. In other embodiments the decoder doesnot need to be part of a transmission system, rather, it may be astandalone device for decoding encoded audio signals or it may be partof an audio signal receiver.

In the following, embodiments of the inventive approach that may beimplemented in at least one of the encoding processor 206 and thedecoding processor 256 will be described in further detail. FIG. 4 showsa flow diagram for processing a current frame of the audio signal inaccordance with an embodiment of the inventive approach. The processingof the current frame will be described, and the past frame is assumed tobe already processed with the same technique described below. Inaccordance with the present invention, in step S100 a current frame ofthe audio signal is received. The current frame is filtered in stepS102, for example in a way as described above with regard to FIGS. 1 and2 (see filter block 102). In accordance with the inventive approach, adiscontinuity between the filtered past frame n−1 and the filteredcurrent frame n (see FIG. 1 or 2) will be removed using linearpredictive filtering as is indicated at step S104. In accordance anembodiment the linear predictive filter may be defined as

${A(z)} = \frac{1}{\Sigma_{m = 0}^{M}a_{m}z^{- m}}$with M the filter order and a_(m) the filter coefficients (with a₀=1).This kind of filter is also known as Linear Predictive Coding (LPC). Inaccordance with embodiments the filtered current frame is processed byapplying linear predictive filtering to at least a part of the filteredcurrent frame. The discontinuity may be removed by modifying a beginningportion of the filtered current frame by a signal obtained by linearpredictive filtering a predefined signal with initial states of thelinear predictive coding filter defined on the basis of a last part ofthe past frame. The initial states of the linear predictive codingfilter may be defined on the basis of a last part of the past framefiltered using the set of filter parameters for the current frame. Theinventive approach is advantageous as it does not require filtering thecurrent frame of an audio signal with a filter coefficient that is usedfor the past frame and thereby avoids problems that arise due to themismatch of the filter parameters for the current frame and for the pastframe as they are experienced in the known approaches described abovewith reference to FIG. 2 .

FIG. 5 shows a schematic block diagram for processing a current audioframe of the audio signal in accordance with embodiments of the presentinvention avoiding undesired distortion in the output signal despite theremoval of the discontinuities. In FIG. 5 , the same reference signs asin FIGS. 1 and 2 are used. A current frame n of the audio signal 100 isreceived, each frame of the audio signal 100 having a plurality ofsamples. The current frame n of the audio signal 100 is processed by thefilter block 102. When compared to the known approaches of FIGS. 1 and 2, in accordance with embodiments as described with regard to FIG. 5 ,the filtered current frame is further processed on the basis of ZIRsamples as is schematically shown by block 110. In accordance with anembodiment on the basis of the past frame n−1, and on the basis of anLPC filter the ZIR samples are produced as is schematically shown byblock 112.

The functionality of the processing blocks 110 and 112 will now bedescribed in further detail. FIG. 6 shows a flow diagram representingthe functionality of the processing block 112 for generating the ZIRsamples. As mentioned above, the frames of an audio signal 100 arefiltered with a linear filter H(z) using filter parameters c selected ordetermined for the respective frame. The filter H(z) may be a recursivefiler, e.g., an IIR filter, or it may be a non-recursive filter, e.g., aFIR filter. In the processing block 112 a LPC filter is used which mayor may not be quantized. The LPC filter is of the order M and may beeither estimated on the filtered or non-filtered audio signal or may bethe LPC filter that is also used in an audio codec. In a first stepS200, the M (M=the order of the LPC filter) last samples of the pastframe n−1 are filtered with the filter H(z) using, however, the filterparameters or coefficients c₁ of the current frame n. Step S200 therebyproduces a first portion of filtered signal. In step S202 the M lastsamples of the filtered past frame n−1 (the M last samples of the pastframe filtered using the filter parameters or coefficients c₀ of thepast frame n−1) are subtracted from the first portion of filtered signalprovided by step S200, thereby producing a second portion of filteredsignal. In step S204 the LPC filter having the order M is applied, morespecifically a zero input response (ZIR) of the LPC filter is generatedin step S204 by filtering a frame of zero samples, wherein the initialstates of the filter are equal to the second portion of filteredsignals, thereby generating the ZIR. In accordance with embodiments, theZIR can be windowed such that its amplitude decreases faster to 0.

The ZIR, as described above with regard to FIG. 5 , is applied in theprocessing block 110, the functionality of which is described withreference to the flow diagram of FIG. 7 for the case of using, as thelinear filer H(z), a recursive filter, like an IIR filter. In accordancewith the embodiment described with regard to FIG. 5 , to removediscontinuities between the current frame and the past frame whileavoiding undesired distortions, filtering the current frame n comprisesprocessing (filtering) the current frame n on a sample-by-sample basis,wherein the samples of the beginning portion are treated in accordancewith the inventive approach. To be more specific, M samples of abeginning portion of the current frame n are processed, and at a firststep S300 the variables m is set to 0. In a next step S302, the sample mof the current frame n is filtered using the filter H(z) and the filtercoefficients or parameters c₁ for the current frame n. Thus, other thanin conventional approaches, the current frame, in accordance with theinventive approach, is not filtered using coefficients from the pastframe, but only coefficients from the current frame, which as aconsequence avoids the undesired distortion which exist in conventionalapproaches despite the fact that discontinuities are removed. Step S302yields a filtered sample m, and in step S304 the ZIR samplecorresponding to sample m is subtracted from the filtered sample myielding the corresponding sample of the filtered current frame n. Instep S306 it is determined whether the last sample M of the beginningportion of the current frame n is processed. In case not all M samplesof the beginning portions have been processed, the variable m isincremented and the method steps S302 to S306 are repeated for the nextsample of the current frame n. Once all M samples of the beginningportions have been processed, at step S308 the remaining samples of thecurrent frame n are filtered using the filter parameters of the currentframe c₁, thereby providing the filtered current frame n processed inaccordance with the inventive approach avoiding undesired distortionupon removal of the discontinuities between consecutive frames.

In accordance with another embodiment, the linear filer H(z) is anon-recursive filter, like a FIR filter, and the ZIR, as described abovewith regard to FIG. 5 , is applied in the processing block 110. Thefunctionality of this embodiment is described with reference to the flowdiagram of FIG. 8 . The current frame n, at step S400, is filtered withthe filter H(z) using the filter coefficients or parameters c₁ for thecurrent frame. Thus, other than in conventional approaches, the currentframe, in accordance with the inventive approach, is not filtered usingcoefficients from the past frame, but only coefficients from the currentframe, which as a consequence avoids the undesired distortion whichexist in conventional approaches despite the fact that discontinuitiesare removed. In step S402 a beginning portion of the ZIR is subtractedfrom a corresponding beginning portion of the filtered current frame,thereby providing the filtered current frame n having the beginningportion filtered/processed in accordance with the inventive approach andthe remaining part only filtered using filter coefficients or parametersc₁ for the current frame, thereby avoiding undesired distortion uponremoval of the discontinuities between consecutive frames.

The inventive approach may be applied in situations as described abovewhen the audio signal is filtered. In accordance with embodiments, theinventive approach may also be applied at the decoder side, for example,when using an audio codec postfilter for reducing the level of codingnoise between signal harmonics. For processing the audio frames at thedecoder the postfilter, in accordance with an embodiment, may be asfollows:H(z)=(1−B(z))/(1−A(z)·z ^(−T))

where B(z) and A(z) are two FIR filters and the H(z) filter parametersare the coefficients of the FIR filters B(z) and A(z), and T indicatesthe pitch lag. In such a scenario, the filter may also introduce adiscontinuity between the two filtered frames, for example when the pastfilter frame parameters c₀ are different from the current frame filterparameters c₁, and such a discontinuity may produce an artifact in thefiltered audio signal 104, for example a “click”. This discontinuity isremoved by processing the filtered current frame as described above indetail.

Although some aspects of the described concept have been described inthe context of an apparatus, it is clear that these aspects alsorepresent a description of the corresponding method, where a block ordevice corresponds to a method step or a feature of a method step.Analogously, aspects described in the context of a method step alsorepresent a description of a corresponding block or item or feature of acorresponding apparatus.

Depending on certain implementation requirements, embodiments of theinvention can be implemented in hardware or in software. Theimplementation can be performed using a digital storage medium, forexample a floppy disk, a DVD, a Blue-Ray, a CD, a ROM, a PROM, an EPROM,an EEPROM or a FLASH memory, having electronically readable controlsignals stored thereon, which cooperate (or are capable of cooperating)with a programmable computer system such that the respective method isperformed. Therefore, the digital storage medium may be computerreadable.

Some embodiments according to the invention comprise a data carrierhaving electronically readable control signals, which are capable ofcooperating with a programmable computer system, such that one of themethods described herein is performed.

Generally, embodiments of the present invention can be implemented as acomputer program product with a program code, the program code beingoperative for performing one of the methods when the computer programproduct runs on a computer. The program code may for example be storedon a machine readable carrier.

Other embodiments comprise the computer program for performing one ofthe methods described herein, stored on a machine readable carrier.

In other words, an embodiment of the inventive method is, therefore, acomputer program having a program code for performing one of the methodsdescribed herein, when the computer program runs on a computer.

A further embodiment of the inventive methods is, therefore, a datacarrier (or a digital storage medium, or a computer-readable medium)comprising, recorded thereon, the computer program for performing one ofthe methods described herein.

A further embodiment of the inventive method is, therefore, a datastream or a sequence of signals representing the computer program forperforming one of the methods described herein. The data stream or thesequence of signals may for example be configured to be transferred viaa data communication connection, for example via the Internet.

A further embodiment comprises a processing means, for example acomputer, or a programmable logic device, configured to or adapted toperform one of the methods described herein.

A further embodiment comprises a computer having installed thereon thecomputer program for performing one of the methods described herein.

In some embodiments, a programmable logic device (for example a fieldprogrammable gate array) may be used to perform some or all of thefunctionalities of the methods described herein. In some embodiments, afield programmable gate array may cooperate with a microprocessor inorder to perform one of the methods described herein. Generally, themethods may be performed by any hardware apparatus.

While this invention has been described in terms of several embodiments,there are alterations, permutations, and equivalents which will beapparent to others skilled in the art and which fall within the scope ofthis invention. It should also be noted that there are many alternativeways of implementing the methods and compositions of the presentinvention. It is therefore intended that the following appended claimsbe interpreted as including all such alterations, permutations, andequivalents as fall within the true spirit and scope of the presentinvention.

What is claimed is:
 1. A method for decoding an audio signal, the method comprising: receiving the audio signal; and decoding the audio signal; wherein decoding the audio signal comprises processing consecutive audio frames of the audio signal to remove a discontinuity by linear predictive filtering, the discontinuity being between a filtered past frame and a filtered current frame of the audio signal, the filtered past frame being a past frame filtered using a set of past filter frame parameters, the filtered current frame being a current frame filtered using a set of current frame filter parameters, and the past filter frame parameters being different from the current frame filter parameters, and the processing comprising: acquiring, by a decoder, a signal by linear predictive filtering a predefined signal with initial states of a linear predictive filter, the initial states defined on basis of a last part of an unfiltered past frame filtered using the set of current frame filter parameters for filtering the current frame, filtering the current frame of the audio signal, and removing the discontinuity by modifying a beginning portion of the filtered current frame by the signal acquired by linear predictive filtering, wherein the linear predictive filter is defined as A(z)=1/(Σ_(m=0) ^(M) a _(m) z ^(−m)) with M the filter order and a_(m) the filter coefficients, with a₀=1, and wherein decoding the audio signal comprises filtering the audio frames using the following filter: H(z)=(1−B(z))/(1−C(z)·z ^(−T)) where B(z) and C(z) are two FIR filters and the H(z) filter parameters are the coefficients of the FIR filters B(z) and C(z), and T indicates the pitch lag.
 2. The method of claim 1, further comprising estimating the linear predictive filter on the filtered or non-filtered audio signal.
 3. The method of claim 2, wherein estimating the linear predictive filter comprises estimating the filter based on the past and/or current frame of the audio signal or based on the past filtered frame of the audio signal using the Levinson-Durbin algorithm.
 4. The method of claim 1, wherein the linear predictive filter comprises a linear predictive filter of an audio codec.
 5. A non-transitory digital storage medium having stored thereon a computer program for performing, when executed by a computer, a method for decoding an audio signal, the method comprising: receiving the audio signal; and decoding the audio signal; wherein decoding the audio signal comprises processing consecutive audio frames of the audio signal to remove a discontinuity by linear predictive filtering, the discontinuity being between a filtered past frame and a filtered current frame of the audio signal, the filtered past frame being a past frame filtered using a set of past filter frame parameters, the filtered current frame being a current frame filtered using a set of current frame filter parameters, and the past filter frame parameters being different from the current frame filter parameters, and the processing comprising: acquiring, by a decoder, a signal by linear predictive filtering a predefined signal with initial states of a linear predictive filter, the initial states defined on basis of a last part of an unfiltered past frame filtered using the set of current frame filter parameters for filtering the current frame, filtering the current frame of the audio signal, and removing the discontinuity by modifying a beginning portion of the filtered current frame by the signal acquired by linear predictive filtering, wherein the linear predictive filter is defined as A(z)=1/(Σ_(m=0) ^(M) a _(m) z ^(−m)) with M the filter order and a_(m), the filter coefficients, with a₀=1, and wherein decoding the audio signal comprises filtering the audio frames using the following filter: H(z)=(1−B(z))/(1−C(z)·z ^(−T)) where B(z) and C(z) are two FIR filters and the H(z) filter parameters are the coefficients of the FIR filters B(z) and C(z), and T indicates the pitch lag.
 6. A decoder for decoding an audio signal, the apparatus comprising: an input for receiving the audio signal; and decoding the audio signal; a processor configured to decode the audio signal, wherein, for decoding the audio signal, the processor is configured to decode consecutive audio frames of the audio signal to remove a discontinuity by linear predictive filtering, the discontinuity being between a filtered past frame and a filtered current frame of the audio signal, the filtered past frame being a past frame filtered using a set of past filter frame parameters, the filtered current frame being a current frame filtered using a set of current frame filter parameters, and the past filter frame parameters being different from the current frame filter parameters, wherein the processor is configured to acquire a signal by linear predictive filtering a predefined signal with initial states of a linear predictive filter, the initial states defined on basis of a last part of an unfiltered past frame filtered using the set of current frame filter parameters for filtering the current frame, filter the current frame of the audio signal, and remove the discontinuity by modifying a beginning portion of the filtered current frame by the signal acquired by linear predictive filtering, wherein the linear predictive filter is defined as A(z)=1/(Σ_(m=0) ^(M) a _(m) z ^(−m)) with M the filter order and a_(m) the filter coefficients, with a₀=1, and wherein decoding the audio signal comprises filtering the audio frames using the following filter: H(z)=(1−B(z))/(1−C(z)·z ^(−T)) where B(z) and C(z) are two FIR filters and the H(z) filter parameters are the coefficients of the FIR filters B(z) and C(z), and T indicates the pitch lag. 